Displaying 5 results from an estimated 5 matches for "npdi".
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npd
2016 Mar 18
2
Incoming INVITE with Portability Info and LRN
...sk's own Location Routing Number (LRN),
> > validating the "rn" in the INVITE and then using the Called Number from
> > the INVITE as the extension in the dialplan.
> >
> > The INVITE R-URI looks like:
> > INVITE
> > sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200
> :5060;user=phone;transport=udp
> > SIP/2.0
> >
> > The +1913663000 is the LRN of the Asterisk box, so I would want to have
> > the dialplan validate that the "rn" is that number. The +19136631291 is
> > the extension within the system tha...
2016 Mar 18
2
Incoming INVITE with Portability Info and LRN
...er so that it will recognize an
incoming call to the Asterisk's own Location Routing Number (LRN),
validating the "rn" in the INVITE and then using the Called Number from the
INVITE as the extension in the dialplan.
The INVITE R-URI looks like:
INVITE sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp
SIP/2.0
The +1913663000 is the LRN of the Asterisk box, so I would want to have the
dialplan validate that the "rn" is that number. The +19136631291 is the
extension within the system that they are trying to reach, that extension
will vary, a...
2014 Jul 10
0
PJSIP Transfer not working
...he call is closed.
Here is a trace. How do I do this?
[Jul 9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869
pbx_extension_helper: Launching 'Transfer'
-- Executing [17274428141 at redirect:30]
Transfer("PJSIP/Client.1.1.1.1-00000002",
"PJSIP/17274428141;rn=+18134029999;npdi at 1.1.1.1") in new stack
[Jul 9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869
pbx_extension_helper: Launching 'Verbose'
-- Executing [17274428141 at redirect:31]
Verbose("PJSIP/Client.1.1.1.1-00000002", "2,Transferred:
17274428141;rn=+18134029999;npdi at 1.1.1.1&q...
2016 Jun 29
2
what is a SIP invite, and who can issue them?
...ia: SIP/2.0/UDP
172.16.1.12;branch=z9hG4bKfae8cb69f547b8cb;received=172.16.0.179
From: <sip:5555555555 at 172.16.1.12>;tag=102
To: <sip:5555555555 at 174.36.199.131>
call-id: 0704037283648236478326200101 at 172.16.1.12
CSeq: 1 INVITE
Contact: Transfer <sip:5555555555;*rn=+15555555556;npdi;*@174.36.199.131>
Content-Length: 0
If a number has been ported the response will contain the dip indicator
("npdi;") as well as the LRN (rn=+1..), otherwise these fields will be
missing
from https://apidocs.telnyx.com/
and then clicking "Data API" and then "SIP reque...
2013 Jan 04
0
T38MaxBitRate issue on fax passthrough
...NVITE sip:4803836933 at XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK7ef2185a;rport
Max-Forwards: 70
From: <sip:6024667281 at YYY.YYY.YYY.YYY>;tag=as40d4ca92
To: <sip:4803836933 at XXX.XXX.XXX.XXX;isup-oli=0>;tag=gK020b0efc
Contact: <sip:6024667281;npdi=yes at ZZZ.ZZZ.ZZZ.ZZZ:5060>
Call-ID: 335684047_12245518 at XXX.XXX.XXX.XXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.5.0-digiumphones
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
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